java使用Sonic 算法对音频变速不变声变调调整音量
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依赖库：https://github.com/waywardgeek/sonic
基础库：Sonic.java
/* Sonic library Copyright 2010, 2011 Bill Cox This file is part of the Sonic Library. This file is licensed under the Apache 2.0 license. */ public class Sonic { private static final int SONIC_MIN_PITCH = 65; private static final int SONIC_MAX_PITCH = 400; // This is used to down-sample some inputs to improve speed private static final int SONIC_AMDF_FREQ = 4000; // The number of points to use in the sinc FIR filter for resampling. private static final int SINC_FILTER_POINTS = 12; private static final int SINC_TABLE_SIZE = 601; // Lookup table for windowed sinc function of SINC_FILTER_POINTS points. private static final short sincTable[] = { 0, 0, 0, 0, 0, 0, 0, -1, -1, -2, -2, -3, -4, -6, -7, -9, -10, -12, -14, -17, -19, -21, -24, -26, -29, -32, -34, -37, -40, -42, -44, -47, -48, -50, -51, -52, -53, -53, -53, -52, -50, -48, -46, -43, -39, -34, -29, -22, -16, -8, 0, 9, 19, 29, 41, 53, 65, 79, 92, 107, 121, 137, 152, 168, 184, 200, 215, 231, 247, 262, 276, 291, 304, 317, 328, 339, 348, 357, 363, 369, 372, 374, 375, 373, 369, 363, 355, 345, 332, 318, 300, 281, 259, 234, 208, 178, 147, 113, 77, 39, 0, -41, -85, -130, -177, -225, -274, -324, -375, -426, -478, -530, -581, -632, -682, -731, -779, -825, -870, -912, -951, -989, -1023, -1053, -1080, -1104, -1123, -1138, -1149, -1154, -1155, -1151, -1141, -1125, -1105, -1078, -1046, -1007, -963, -913, -857, -796, -728, -655, -576, -492, -403, -309, -210, -107, 0, 111, 225, 342, 462, 584, 708, 833, 958, 1084, 1209, 1333, 1455, 1575, 1693, 1807, 1916, 2022, 2122, 2216, 2304, 2384, 2457, 2522, 2579, 2625, 2663, 2689, 2706, 2711, 2705, 2687, 2657, 2614, 2559, 2491, 2411, 2317, 2211, 2092, 1960, 1815, 1658, 1489, 1308, 1115, 912, 698, 474, 241, 0, -249, -506, -769, -1037, -1310, -1586, -1864, -2144, -2424, -2703, -2980, -3254, -3523, -3787, -4043, -4291, -4529, -4757, -4972, -5174, -5360, -5531, -5685, -5819, -5935, -6029, -6101, -6150, -6175, -6175, -6149, -6096, -6015, -5905, -5767, -5599, -5401, -5172, -4912, -4621, -4298, -3944, -3558, -3141, -2693, -2214, -1705, -1166, -597, 0, 625, 1277, 1955, 2658, 3386, 4135, 4906, 5697, 6506, 7332, 8173, 9027, 9893, 10769, 11654, 12544, 13439, 14335, 15232, 16128, 17019, 17904, 18782, 19649, 20504, 21345, 22170, 22977, 23763, 24527, 25268, 25982, 26669, 27327, 27953, 28547, 29107, 29632, 30119, 30569, 30979, 31349, 31678, 31964, 32208, 32408, 32565, 32677, 32744, 32767, 32744, 32677, 32565, 32408, 32208, 31964, 31678, 31349, 30979, 30569, 30119, 29632, 29107, 28547, 27953, 27327, 26669, 25982, 25268, 24527, 23763, 22977, 22170, 21345, 20504, 19649, 18782, 17904, 17019, 16128, 15232, 14335, 13439, 12544, 11654, 10769, 9893, 9027, 8173, 7332, 6506, 5697, 4906, 4135, 3386, 2658, 1955, 1277, 625, 0, -597, -1166, -1705, -2214, -2693, -3141, -3558, -3944, -4298, -4621, -4912, -5172, -5401, -5599, -5767, -5905, -6015, -6096, -6149, -6175, -6175, -6150, -6101, -6029, -5935, -5819, -5685, -5531, -5360, -5174, -4972, -4757, -4529, -4291, -4043, -3787, -3523, -3254, -2980, -2703, -2424, -2144, -1864, -1586, -1310, -1037, -769, -506, -249, 0, 241, 474, 698, 912, 1115, 1308, 1489, 1658, 1815, 1960, 2092, 2211, 2317, 2411, 2491, 2559, 2614, 2657, 2687, 2705, 2711, 2706, 2689, 2663, 2625, 2579, 2522, 2457, 2384, 2304, 2216, 2122, 2022, 1916, 1807, 1693, 1575, 1455, 1333, 1209, 1084, 958, 833, 708, 584, 462, 342, 225, 111, 0, -107, -210, -309, -403, -492, -576, -655, -728, -796, -857, -913, -963, -1007, -1046, -1078, -1105, -1125, -1141, -1151, -1155, -1154, -1149, -1138, -1123, -1104, -1080, -1053, -1023, -989, -951, -912, -870, -825, -779, -731, -682, -632, -581, -530, -478, -426, -375, -324, -274, -225, -177, -130, -85, -41, 0, 39, 77, 113, 147, 178, 208, 234, 259, 281, 300, 318, 332, 345, 355, 363, 369, 373, 375, 374, 372, 369, 363, 357, 348, 339, 328, 317, 304, 291, 276, 262, 247, 231, 215, 200, 184, 168, 152, 137, 121, 107, 92, 79, 65, 53, 41, 29, 19, 9, 0, -8, -16, -22, -29, -34, -39, -43, -46, -48, -50, -52, -53, -53, -53, -52, -51, -50, -48, -47, -44, -42, -40, -37, -34, -32, -29, -26, -24, -21, -19, -17, -14, -12, -10, -9, -7, -6, -4, -3, -2, -2, -1, -1, 0, 0, 0, 0, 0, 0, 0 }; private short inputBuffer[]; private short outputBuffer[]; private short pitchBuffer[]; private short downSampleBuffer[]; private float speed; private float volume; private float pitch; private float rate; private int oldRatePosition; private int newRatePosition; private boolean useChordPitch; private int quality; private int numChannels; private int inputBufferSize; private int pitchBufferSize; private int outputBufferSize; private int numInputSamples; private int numOutputSamples; private int numPitchSamples; private int minPeriod; private int maxPeriod; private int maxRequired; private int remainingInputToCopy; private int sampleRate; private int prevPeriod; private int prevMinDiff; private int minDiff; private int maxDiff; // Resize the array. private short[] resize( short[] oldArray, int newLength) { newLength *= numChannels; short[] newArray = new short[newLength]; int length = oldArray.length <= newLength? oldArray.length : newLength; System.arraycopy(oldArray, 0, newArray, 0, length); return newArray; } // Move samples from one array to another. May move samples down within an array, but not up. private void move( short dest[], int destPos, short source[], int sourcePos, int numSamples) { System.arraycopy(source, sourcePos*numChannels, dest, destPos*numChannels, numSamples*numChannels); } // Scale the samples by the factor. private void scaleSamples( short samples[], int position, int numSamples, float volume) { int fixedPointVolume = (int)(volume*4096.0f); int start = position*numChannels; int stop = start + numSamples*numChannels; for(int xSample = start; xSample < stop; xSample++) { int value = (samples[xSample]*fixedPointVolume) >> 12; if(value > 32767) { value = 32767; } else if(value < -32767) { value = -32767; } samples[xSample] = (short)value; } } // Get the speed of the stream. public float getSpeed() { return speed; } // Set the speed of the stream. public void setSpeed( float speed) { this.speed = speed; } // Get the pitch of the stream. public float getPitch() { return pitch; } // Set the pitch of the stream. public void setPitch( float pitch) { this.pitch = pitch; } // Get the rate of the stream. public float getRate() { return rate; } // Set the playback rate of the stream. This scales pitch and speed at the same time. public void setRate( float rate) { this.rate = rate; this.oldRatePosition = 0; this.newRatePosition = 0; } // Get the vocal chord pitch setting. public boolean getChordPitch() { return useChordPitch; } // Set the vocal chord mode for pitch computation. Default is off. public void setChordPitch( boolean useChordPitch) { this.useChordPitch = useChordPitch; } // Get the quality setting. public int getQuality() { return quality; } // Set the "quality". Default 0 is virtually as good as 1, but very much faster. public void setQuality( int quality) { this.quality = quality; } // Get the scaling factor of the stream. public float getVolume() { return volume; } // Set the scaling factor of the stream. public void setVolume( float volume) { this.volume = volume; } // Allocate stream buffers. private void allocateStreamBuffers( int sampleRate, int numChannels) { minPeriod = sampleRate/SONIC_MAX_PITCH; maxPeriod = sampleRate/SONIC_MIN_PITCH; maxRequired = 2*maxPeriod; inputBufferSize = maxRequired; inputBuffer = new short[maxRequired*numChannels]; outputBufferSize = maxRequired; outputBuffer = new short[maxRequired*numChannels]; pitchBufferSize = maxRequired; pitchBuffer = new short[maxRequired*numChannels]; downSampleBuffer = new short[maxRequired]; this.sampleRate = sampleRate; this.numChannels = numChannels; oldRatePosition = 0; newRatePosition = 0; prevPeriod = 0; } // Create a sonic stream. public Sonic( int sampleRate, int numChannels) { allocateStreamBuffers(sampleRate, numChannels); speed = 1.0f; pitch = 1.0f; volume = 1.0f; rate = 1.0f; oldRatePosition = 0; newRatePosition = 0; useChordPitch = false; quality = 0; } // Get the sample rate of the stream. public int getSampleRate() { return sampleRate; } // Set the sample rate of the stream. This will cause samples buffered in the stream to be lost. public void setSampleRate( int sampleRate) { allocateStreamBuffers(sampleRate, numChannels); } // Get the number of channels. public int getNumChannels() { return numChannels; } // Set the num channels of the stream. This will cause samples buffered in the stream to be lost. public void setNumChannels( int numChannels) { allocateStreamBuffers(sampleRate, numChannels); } // Enlarge the output buffer if needed. private void enlargeOutputBufferIfNeeded( int numSamples) { if(numOutputSamples + numSamples > outputBufferSize) { outputBufferSize += (outputBufferSize >> 1) + numSamples; outputBuffer = resize(outputBuffer, outputBufferSize); } } // Enlarge the input buffer if needed. private void enlargeInputBufferIfNeeded( int numSamples) { if(numInputSamples + numSamples > inputBufferSize) { inputBufferSize += (inputBufferSize >> 1) + numSamples; inputBuffer = resize(inputBuffer, inputBufferSize); } } // Add the input samples to the input buffer. private void addFloatSamplesToInputBuffer( float samples[], int numSamples) { if(numSamples == 0) { return; } enlargeInputBufferIfNeeded(numSamples); int xBuffer = numInputSamples*numChannels; for(int xSample = 0; xSample < numSamples*numChannels; xSample++) { inputBuffer[xBuffer++] = (short)(samples[xSample]*32767.0f); } numInputSamples += numSamples; } // Add the input samples to the input buffer. private void addShortSamplesToInputBuffer( short samples[], int numSamples) { if(numSamples == 0) { return; } enlargeInputBufferIfNeeded(numSamples); move(inputBuffer, numInputSamples, samples, 0, numSamples); numInputSamples += numSamples; } // Add the input samples to the input buffer. private void addUnsignedByteSamplesToInputBuffer( byte samples[], int numSamples) { short sample; enlargeInputBufferIfNeeded(numSamples); int xBuffer = numInputSamples*numChannels; for(int xSample = 0; xSample < numSamples*numChannels; xSample++) { sample = (short)((samples[xSample] & 0xff) - 128); // Convert from unsigned to signed inputBuffer[xBuffer++] = (short) (sample << 8); } numInputSamples += numSamples; } // Add the input samples to the input buffer. They must be 16-bit little-endian encoded in a byte array. private void addBytesToInputBuffer( byte inBuffer[], int numBytes) { int numSamples = numBytes/(2*numChannels); short sample; enlargeInputBufferIfNeeded(numSamples); int xBuffer = numInputSamples*numChannels; for(int xByte = 0; xByte + 1 < numBytes; xByte += 2) { sample = (short)((inBuffer[xByte] & 0xff) | (inBuffer[xByte + 1] << 8)); inputBuffer[xBuffer++] = sample; } numInputSamples += numSamples; } // Remove input samples that we have already processed. private void removeInputSamples( int position) { int remainingSamples = numInputSamples - position; move(inputBuffer, 0, inputBuffer, position, remainingSamples); numInputSamples = remainingSamples; } // Just copy from the array to the output buffer private void copyToOutput( short samples[], int position, int numSamples) { enlargeOutputBufferIfNeeded(numSamples); move(outputBuffer, numOutputSamples, samples, position, numSamples); numOutputSamples += numSamples; } // Just copy from the input buffer to the output buffer. Return num samples copied. private int copyInputToOutput( int position) { int numSamples = remainingInputToCopy; if(numSamples > maxRequired) { numSamples = maxRequired; } copyToOutput(inputBuffer, position, numSamples); remainingInputToCopy -= numSamples; return numSamples; } // Read data out of the stream. Sometimes no data will be available, and zero // is returned, which is not an error condition. public int readFloatFromStream( float samples[], int maxSamples) { int numSamples = numOutputSamples; int remainingSamples = 0; if(numSamples == 0) { return 0; } if(numSamples > maxSamples) { remainingSamples = numSamples - maxSamples; numSamples = maxSamples; } for(int xSample = 0; xSample < numSamples*numChannels; xSample++) { samples[xSample] = (outputBuffer[xSample])/32767.0f; } move(outputBuffer, 0, outputBuffer, numSamples, remainingSamples); numOutputSamples = remainingSamples; return numSamples; } // Read short data out of the stream. Sometimes no data will be available, and zero // is returned, which is not an error condition. public int readShortFromStream( short samples[], int maxSamples) { int numSamples = numOutputSamples; int remainingSamples = 0; if(numSamples == 0) { return 0; } if(numSamples > maxSamples) { remainingSamples = numSamples - maxSamples; numSamples = maxSamples; } move(samples, 0, outputBuffer, 0, numSamples); move(outputBuffer, 0, outputBuffer, numSamples, remainingSamples); numOutputSamples = remainingSamples; return numSamples; } // Read unsigned byte data out of the stream. Sometimes no data will be available, and zero // is returned, which is not an error condition. public int readUnsignedByteFromStream( byte samples[], int maxSamples) { int numSamples = numOutputSamples; int remainingSamples = 0; if(numSamples == 0) { return 0; } if(numSamples > maxSamples) { remainingSamples = numSamples - maxSamples; numSamples = maxSamples; } for(int xSample = 0; xSample < numSamples*numChannels; xSample++) { samples[xSample] = (byte)((outputBuffer[xSample] >> 8) + 128); } move(outputBuffer, 0, outputBuffer, numSamples, remainingSamples); numOutputSamples = remainingSamples; return numSamples; } // Read unsigned byte data out of the stream. Sometimes no data will be available, and zero // is returned, which is not an error condition. public int readBytesFromStream( byte outBuffer[], int maxBytes) { int maxSamples = maxBytes/(2*numChannels); int numSamples = numOutputSamples; int remainingSamples = 0; if(numSamples == 0 || maxSamples == 0) { return 0; } if(numSamples > maxSamples) { remainingSamples = numSamples - maxSamples; numSamples = maxSamples; } for(int xSample = 0; xSample < numSamples*numChannels; xSample++) { short sample = outputBuffer[xSample]; outBuffer[xSample << 1] = (byte)(sample & 0xff); outBuffer[(xSample << 1) + 1] = (byte)(sample >> 8); } move(outputBuffer, 0, outputBuffer, numSamples, remainingSamples); numOutputSamples = remainingSamples; return 2*numSamples*numChannels; } // Force the sonic stream to generate output using whatever data it currently // has. No extra delay will be added to the output, but flushing in the middle of // words could introduce distortion. public void flushStream() { int remainingSamples = numInputSamples; float s = speed/pitch; float r = rate*pitch; int expectedOutputSamples = numOutputSamples + (int)((remainingSamples/s + numPitchSamples)/r + 0.5f); // Add enough silence to flush both input and pitch buffers. enlargeInputBufferIfNeeded(remainingSamples + 2*maxRequired); for(int xSample = 0; xSample < 2*maxRequired*numChannels; xSample++) { inputBuffer[remainingSamples*numChannels + xSample] = 0; } numInputSamples += 2*maxRequired; writeShortToStream(null, 0); // Throw away any extra samples we generated due to the silence we added. if(numOutputSamples > expectedOutputSamples) { numOutputSamples = expectedOutputSamples; } // Empty input and pitch buffers. numInputSamples = 0; remainingInputToCopy = 0; numPitchSamples = 0; } // Return the number of samples in the output buffer public int samplesAvailable() { return numOutputSamples; } // If skip is greater than one, average skip samples together and write them to // the down-sample buffer. If numChannels is greater than one, mix the channels // together as we down sample. private void downSampleInput( short samples[], int position, int skip) { int numSamples = maxRequired/skip; int samplesPerValue = numChannels*skip; int value; position *= numChannels; for(int i = 0; i < numSamples; i++) { value = 0; for(int j = 0; j < samplesPerValue; j++) { value += samples[position + i*samplesPerValue + j]; } value /= samplesPerValue; downSampleBuffer[i] = (short)value; } } // Find the best frequency match in the range, and given a sample skip multiple. // For now, just find the pitch of the first channel. private int findPitchPeriodInRange( short samples[], int position, int minPeriod, int maxPeriod) { int bestPeriod = 0, worstPeriod = 255; int minDiff = 1, maxDiff = 0; position *= numChannels; for(int period = minPeriod; period <= maxPeriod; period++) { int diff = 0; for(int i = 0; i < period; i++) { short sVal = samples[position + i]; short pVal = samples[position + period + i]; diff += sVal >= pVal? sVal - pVal : pVal - sVal; } /* Note that the highest number of samples we add into diff will be less than 256, since we skip samples. Thus, diff is a 24 bit number, and we can safely multiply by numSamples without overflow */ if(diff*bestPeriod < minDiff*period) { minDiff = diff; bestPeriod = period; } if(diff*worstPeriod > maxDiff*period) { maxDiff = diff; worstPeriod = period; } } this.minDiff = minDiff/bestPeriod; this.maxDiff = maxDiff/worstPeriod; return bestPeriod; } // At abrupt ends of voiced words, we can have pitch periods that are better // approximated by the previous pitch period estimate. Try to detect this case. private boolean prevPeriodBetter( int minDiff, int maxDiff, boolean preferNewPeriod) { if(minDiff == 0 || prevPeriod == 0) { return false; } if(preferNewPeriod) { if(maxDiff > minDiff*3) { // Got a reasonable match this period return false; } if(minDiff*2 <= prevMinDiff*3) { // Mismatch is not that much greater this period return false; } } else { if(minDiff <= prevMinDiff) { return false; } } return true; } // Find the pitch period. This is a critical step, and we may have to try // multiple ways to get a good answer. This version uses AMDF. To improve // speed, we down sample by an integer factor get in the 11KHz range, and then // do it again with a narrower frequency range without down sampling private int findPitchPeriod( short samples[], int position, boolean preferNewPeriod) { int period, retPeriod; int skip = 1; if(sampleRate > SONIC_AMDF_FREQ && quality == 0) { skip = sampleRate/SONIC_AMDF_FREQ; } if(numChannels == 1 && skip == 1) { period = findPitchPeriodInRange(samples, position, minPeriod, maxPeriod); } else { downSampleInput(samples, position, skip); period = findPitchPeriodInRange(downSampleBuffer, 0, minPeriod/skip, maxPeriod/skip); if(skip != 1) { period *= skip; int minP = period - (skip << 2); int maxP = period + (skip << 2); if(minP < minPeriod) { minP = minPeriod; } if(maxP > maxPeriod) { maxP = maxPeriod; } if(numChannels == 1) { period = findPitchPeriodInRange(samples, position, minP, maxP); } else { downSampleInput(samples, position, 1); period = findPitchPeriodInRange(downSampleBuffer, 0, minP, maxP); } } } if(prevPeriodBetter(minDiff, maxDiff, preferNewPeriod)) { retPeriod = prevPeriod; } else { retPeriod = period; } prevMinDiff = minDiff; prevPeriod = period; return retPeriod; } // Overlap two sound segments, ramp the volume of one down, while ramping the // other one from zero up, and add them, storing the result at the output. private void overlapAdd( int numSamples, int numChannels, short out[], int outPos, short rampDown[], int rampDownPos, short rampUp[], int rampUpPos) { for(int i = 0; i < numChannels; i++) { int o = outPos*numChannels + i; int u = rampUpPos*numChannels + i; int d = rampDownPos*numChannels + i; for(int t = 0; t < numSamples; t++) { out[o] = (short)((rampDown[d]*(numSamples - t) + rampUp[u]*t)/numSamples); o += numChannels; d += numChannels; u += numChannels; } } } // Overlap two sound segments, ramp the volume of one down, while ramping the // other one from zero up, and add them, storing the result at the output. private void overlapAddWithSeparation( int numSamples, int numChannels, int separation, short out[], int outPos, short rampDown[], int rampDownPos, short rampUp[], int rampUpPos) { for(int i = 0; i < numChannels; i++) { int o = outPos*numChannels + i; int u = rampUpPos*numChannels + i; int d = rampDownPos*numChannels + i; for(int t = 0; t < numSamples + separation; t++) { if(t < separation) { out[o] = (short)(rampDown[d]*(numSamples - t)/numSamples); d += numChannels; } else if(t < numSamples) { out[o] = (short)((rampDown[d]*(numSamples - t) + rampUp[u]*(t - separation))/numSamples); d += numChannels; u += numChannels; } else { out[o] = (short)(rampUp[u]*(t - separation)/numSamples); u += numChannels; } o += numChannels; } } } // Just move the new samples in the output buffer to the pitch buffer private void moveNewSamplesToPitchBuffer( int originalNumOutputSamples) { int numSamples = numOutputSamples - originalNumOutputSamples; if(numPitchSamples + numSamples > pitchBufferSize) { pitchBufferSize += (pitchBufferSize >> 1) + numSamples; pitchBuffer = resize(pitchBuffer, pitchBufferSize); } move(pitchBuffer, numPitchSamples, outputBuffer, originalNumOutputSamples, numSamples); numOutputSamples = originalNumOutputSamples; numPitchSamples += numSamples; } // Remove processed samples from the pitch buffer. private void removePitchSamples( int numSamples) { if(numSamples == 0) { return; } move(pitchBuffer, 0, pitchBuffer, numSamples, numPitchSamples - numSamples); numPitchSamples -= numSamples; } // Change the pitch. The latency this introduces could be reduced by looking at // past samples to determine pitch, rather than future. private void adjustPitch( int originalNumOutputSamples) { int period, newPeriod, separation; int position = 0; if(numOutputSamples == originalNumOutputSamples) { return; } moveNewSamplesToPitchBuffer(originalNumOutputSamples); while(numPitchSamples - position >= maxRequired) { period = findPitchPeriod(pitchBuffer, position, false); newPeriod = (int)(period/pitch); enlargeOutputBufferIfNeeded(newPeriod); if(pitch >= 1.0f) { overlapAdd(newPeriod, numChannels, outputBuffer, numOutputSamples, pitchBuffer, position, pitchBuffer, position + period - newPeriod); } else { separation = newPeriod - period; overlapAddWithSeparation(period, numChannels, separation, outputBuffer, numOutputSamples, pitchBuffer, position, pitchBuffer, position); } numOutputSamples += newPeriod; position += period; } removePitchSamples(position); } // Aproximate the sinc function times a Hann window from the sinc table. private int findSincCoefficient(int i, int ratio, int width) { int lobePoints = (SINC_TABLE_SIZE-1)/SINC_FILTER_POINTS; int left = i*lobePoints + (ratio*lobePoints)/width; int right = left + 1; int position = i*lobePoints*width + ratio*lobePoints - left*width; int leftVal = sincTable[left]; int rightVal = sincTable[right]; return ((leftVal*(width - position) + rightVal*position) << 1)/width; } // Return 1 if value >= 0, else -1. This represents the sign of value. private int getSign(int value) { return value >= 0? 1 : -1; } // Interpolate the new output sample. private short interpolate( short in[], int inPos, // Index to first sample which already includes channel offset. int oldSampleRate, int newSampleRate) { // Compute N-point sinc FIR-filter here. Clip rather than overflow. int i; int total = 0; int position = newRatePosition*oldSampleRate; int leftPosition = oldRatePosition*newSampleRate; int rightPosition = (oldRatePosition + 1)*newSampleRate; int ratio = rightPosition - position - 1; int width = rightPosition - leftPosition; int weight, value; int oldSign; int overflowCount = 0; for (i = 0; i < SINC_FILTER_POINTS; i++) { weight = findSincCoefficient(i, ratio, width); /* printf("%u %f ", i, weight); */ value = in[inPos + i*numChannels]*weight; oldSign = getSign(total); total += value; if (oldSign != getSign(total) && getSign(value) == oldSign) { /* We must have overflowed. This can happen with a sinc filter. */ overflowCount += oldSign; } } /* It is better to clip than to wrap if there was a overflow. */ if (overflowCount > 0) { return Short.MAX_VALUE; } else if (overflowCount < 0) { return Short.MIN_VALUE; } return (short)(total >> 16); } // Change the rate. private void adjustRate( float rate, int originalNumOutputSamples) { int newSampleRate = (int)(sampleRate/rate); int oldSampleRate = sampleRate; int position; // Set these values to help with the integer math while(newSampleRate > (1 << 14) || oldSampleRate > (1 << 14)) { newSampleRate >>= 1; oldSampleRate >>= 1; } if(numOutputSamples == originalNumOutputSamples) { return; } moveNewSamplesToPitchBuffer(originalNumOutputSamples); // Leave at least one pitch sample in the buffer for(position = 0; position < numPitchSamples - 1; position++) { while((oldRatePosition + 1)*newSampleRate > newRatePosition*oldSampleRate) { enlargeOutputBufferIfNeeded(1); for(int i = 0; i < numChannels; i++) { outputBuffer[numOutputSamples*numChannels + i] = interpolate(pitchBuffer, position*numChannels + i, oldSampleRate, newSampleRate); } newRatePosition++; numOutputSamples++; } oldRatePosition++; if(oldRatePosition == oldSampleRate) { oldRatePosition = 0; if(newRatePosition != newSampleRate) { System.out.printf("Assertion failed: newRatePosition != newSampleRate "); assert false; } newRatePosition = 0; } } removePitchSamples(position); } // Skip over a pitch period, and copy period/speed samples to the output private int skipPitchPeriod( short samples[], int position, float speed, int period) { int newSamples; if(speed >= 2.0f) { newSamples = (int)(period/(speed - 1.0f)); } else { newSamples = period; remainingInputToCopy = (int)(period*(2.0f - speed)/(speed - 1.0f)); } enlargeOutputBufferIfNeeded(newSamples); overlapAdd(newSamples, numChannels, outputBuffer, numOutputSamples, samples, position, samples, position + period); numOutputSamples += newSamples; return newSamples; } // Insert a pitch period, and determine how much input to copy directly. private int insertPitchPeriod( short samples[], int position, float speed, int period) { int newSamples; if(speed < 0.5f) { newSamples = (int)(period*speed/(1.0f - speed)); } else { newSamples = period; remainingInputToCopy = (int)(period*(2.0f*speed - 1.0f)/(1.0f - speed)); } enlargeOutputBufferIfNeeded(period + newSamples); move(outputBuffer, numOutputSamples, samples, position, period); overlapAdd(newSamples, numChannels, outputBuffer, numOutputSamples + period, samples, position + period, samples, position); numOutputSamples += period + newSamples; return newSamples; } // Resample as many pitch periods as we have buffered on the input. Return 0 if // we fail to resize an input or output buffer. Also scale the output by the volume. private void changeSpeed( float speed) { int numSamples = numInputSamples; int position = 0, period, newSamples; if(numInputSamples < maxRequired) { return; } do { if(remainingInputToCopy > 0) { newSamples = copyInputToOutput(position); position += newSamples; } else { period = findPitchPeriod(inputBuffer, position, true); if(speed > 1.0) { newSamples = skipPitchPeriod(inputBuffer, position, speed, period); position += period + newSamples; } else { newSamples = insertPitchPeriod(inputBuffer, position, speed, period); position += newSamples; } } } while(position + maxRequired <= numSamples); removeInputSamples(position); } // Resample as many pitch periods as we have buffered on the input. Scale the output by the volume. private void processStreamInput() { int originalNumOutputSamples = numOutputSamples; float s = speed/pitch; float r = rate; if(!useChordPitch) { r *= pitch; } if(s > 1.00001 || s < 0.99999) { changeSpeed(s); } else { copyToOutput(inputBuffer, 0, numInputSamples); numInputSamples = 0; } if(useChordPitch) { if(pitch != 1.0f) { adjustPitch(originalNumOutputSamples); } } else if(r != 1.0f) { adjustRate(r, originalNumOutputSamples); } if(volume != 1.0f) { // Adjust output volume. scaleSamples(outputBuffer, originalNumOutputSamples, numOutputSamples - originalNumOutputSamples, volume); } } // Write floating point data to the input buffer and process it. public void writeFloatToStream( float samples[], int numSamples) { addFloatSamplesToInputBuffer(samples, numSamples); processStreamInput(); } // Write the data to the input stream, and process it. public void writeShortToStream( short samples[], int numSamples) { addShortSamplesToInputBuffer(samples, numSamples); processStreamInput(); } // Simple wrapper around sonicWriteFloatToStream that does the unsigned byte to short // conversion for you. public void writeUnsignedByteToStream( byte samples[], int numSamples) { addUnsignedByteSamplesToInputBuffer(samples, numSamples); processStreamInput(); } // Simple wrapper around sonicWriteBytesToStream that does the byte to 16-bit LE conversion. public void writeBytesToStream( byte inBuffer[], int numBytes) { addBytesToInputBuffer(inBuffer, numBytes); processStreamInput(); } // This is a non-stream oriented interface to just change the speed of a sound sample public static int changeFloatSpeed( float samples[], int numSamples, float speed, float pitch, float rate, float volume, boolean useChordPitch, int sampleRate, int numChannels) { Sonic stream = new Sonic(sampleRate, numChannels); stream.setSpeed(speed); stream.setPitch(pitch); stream.setRate(rate); stream.setVolume(volume); stream.setChordPitch(useChordPitch); stream.writeFloatToStream(samples, numSamples); stream.flushStream(); numSamples = stream.samplesAvailable(); stream.readFloatFromStream(samples, numSamples); return numSamples; } /* This is a non-stream oriented interface to just change the speed of a sound sample */ public int sonicChangeShortSpeed( short samples[], int numSamples, float speed, float pitch, float rate, float volume, boolean useChordPitch, int sampleRate, int numChannels) { Sonic stream = new Sonic(sampleRate, numChannels); stream.setSpeed(speed); stream.setPitch(pitch); stream.setRate(rate); stream.setVolume(volume); stream.setChordPitch(useChordPitch); stream.writeShortToStream(samples, numSamples); stream.flushStream(); numSamples = stream.samplesAvailable(); stream.readShortFromStream(samples, numSamples); return numSamples; } }
1、变速不变音
/* This file was written by Bill Cox in 2011, and is licensed under the Apache 2.0 license. */ import com.google.common.io.ByteArrayDataOutput; import com.google.common.io.ByteStreams; import com.netmarch.web.open.util.Sonic; import com.sun.media.sound.WaveFileWriter; import javax.sound.sampled.AudioFileFormat; import javax.sound.sampled.AudioFormat; import javax.sound.sampled.AudioInputStream; import javax.sound.sampled.Audiosystem; import javax.sound.sampled.DataLine; import javax.sound.sampled.LineUnavailableException; import javax.sound.sampled.SourceDataLine; import javax.sound.sampled.UnsupportedAudioFileException; import java.io.ByteArrayInputStream; import java.io.File; import java.io.FileOutputStream; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; public class TestSonic { static ByteArrayDataOutput out = ByteStreams.newDataOutput(); // Run sonic. private static void runSonic( AudioInputStream audioStream, SourceDataLine line, float speed, float pitch, float rate, float volume, boolean emulateChordPitch, int quality, int sampleRate, int numChannels) throws IOException { Sonic sonic = new Sonic(sampleRate, numChannels); int bufferSize = line.getBufferSize(); byte inBuffer[] = new byte[bufferSize]; byte outBuffer[] = new byte[bufferSize]; int numRead, numWritten; sonic.setSpeed(speed); sonic.setPitch(pitch); sonic.setRate(rate); sonic.setVolume(volume); sonic.setChordPitch(emulateChordPitch); sonic.setQuality(quality); do { numRead = audioStream.read(inBuffer, 0, bufferSize); if(numRead <= 0) { sonic.flushStream(); } else { sonic.writeBytesToStream(inBuffer, numRead); } do { numWritten = sonic.readBytesFromStream(outBuffer, bufferSize); if(numWritten > 0) { line.write(outBuffer, 0, numWritten); byte[] target = new byte[numWritten]; System.arraycopy(outBuffer, 0, target, 0, numWritten); out.write(target); } } while(numWritten > 0); } while(numRead > 0); saveFile(out.toByteArray()); } public static void saveFile(byte[] bytes) { String fileName = "c:/"+ System.currentTimeMillis() + ".wav"; OutputStream outStream = null; try { outStream = new FileOutputStream(new File(fileName)); WaveFileWriter writer = new WaveFileWriter(); AudioFormat frmt = new AudioFormat(16000, 16, 1, true, false); ByteArrayInputStream bi = new ByteArrayInputStream(bytes); InputStream in = new AudioInputStream(bi, frmt,bytes.length); writer.write((AudioInputStream) in, AudioFileFormat.Type.WAVE, outStream); outStream.close(); } catch (Exception e) { e.printStackTrace(); } } public static void main( String[] argv) throws UnsupportedAudioFileException, IOException, LineUnavailableException { float speed = 1.5f;//1.5倍速度播放 float pitch = 1.0f; float rate = 1.0f; float volume = 1.0f; boolean emulateChordPitch = false; int quality = 0; AudioInputStream stream = AudioSystem.getAudioInputStream(new File("C:\Users\Administrator\AppData\Local\Temp\aefee1de-f856-4532-8a1c-79a00c2517dd\generated.wav")); AudioFormat format = stream.getFormat(); int sampleRate = (int)format.getSampleRate(); int numChannels = format.getChannels(); SourceDataLine.Info info = new DataLine.Info(SourceDataLine.class, format, ((int)stream.getFrameLength()*format.getFrameSize())); SourceDataLine line = (SourceDataLine)AudioSystem.getLine(info); line.open(stream.getFormat()); line.start(); runSonic(stream, line, speed, pitch, rate, volume, emulateChordPitch, quality, sampleRate, numChannels); line.drain(); line.stop(); } }
2、调整音量
package com.netmarch; /* This file was written by Bill Cox in 2011, and is licensed under the Apache 2.0 license. */ import com.google.common.io.ByteArrayDataOutput; import com.google.common.io.ByteStreams; import com.netmarch.web.open.util.Sonic; import com.sun.media.sound.WaveFileWriter; import javax.sound.sampled.AudioFileFormat; import javax.sound.sampled.AudioFormat; import javax.sound.sampled.AudioInputStream; import javax.sound.sampled.AudioSystem; import javax.sound.sampled.DataLine; import javax.sound.sampled.LineUnavailableException; import javax.sound.sampled.SourceDataLine; import javax.sound.sampled.UnsupportedAudioFileException; import java.io.ByteArrayInputStream; import java.io.File; import java.io.FileOutputStream; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; public class TestSonic { static ByteArrayDataOutput out = ByteStreams.newDataOutput(); // Run sonic. private static void runSonic( AudioInputStream audioStream, SourceDataLine line, float speed, float pitch, float rate, float volume, boolean emulateChordPitch, int quality, int sampleRate, int numChannels) throws IOException { Sonic sonic = new Sonic(sampleRate, numChannels); int bufferSize = line.getBufferSize(); byte inBuffer[] = new byte[bufferSize]; byte outBuffer[] = new byte[bufferSize]; int numRead, numWritten; sonic.setSpeed(speed); sonic.setPitch(pitch); sonic.setRate(rate); sonic.setVolume(volume); sonic.setChordPitch(emulateChordPitch); sonic.setQuality(quality); do { numRead = audioStream.read(inBuffer, 0, bufferSize); if(numRead <= 0) { sonic.flushStream(); } else { sonic.writeBytesToStream(inBuffer, numRead); } do { numWritten = sonic.readBytesFromStream(outBuffer, bufferSize); if(numWritten > 0) { line.write(outBuffer, 0, numWritten); byte[] target = new byte[numWritten]; System.arraycopy(outBuffer, 0, target, 0, numWritten); out.write(target); } } while(numWritten > 0); } while(numRead > 0); //saveFile(out.toByteArray()); } public static void saveFile(byte[] bytes) { String fileName = "c:/"+ System.currentTimeMillis() + ".wav"; OutputStream outStream = null; try { outStream = new FileOutputStream(new File(fileName)); WaveFileWriter writer = new WaveFileWriter(); AudioFormat frmt = new AudioFormat(16000, 16, 1, true, false); ByteArrayInputStream bi = new ByteArrayInputStream(bytes); InputStream in = new AudioInputStream(bi, frmt,bytes.length); writer.write((AudioInputStream) in, AudioFileFormat.Type.WAVE, outStream); outStream.close(); } catch (Exception e) { e.printStackTrace(); } } public static void main( String[] argv) throws UnsupportedAudioFileException, IOException, LineUnavailableException { float speed = 1.0f; float pitch = 1.0f; float rate = 1.0f; float volume = 0.5f; //调整音量 boolean emulateChordPitch = false; int quality = 0; AudioInputStream stream = AudioSystem.getAudioInputStream(new File("d:\generated.wav")); AudioFormat format = stream.getFormat(); int sampleRate = (int)format.getSampleRate(); int numChannels = format.getChannels(); SourceDataLine.Info info = new DataLine.Info(SourceDataLine.class, format, ((int)stream.getFrameLength()*format.getFrameSize())); SourceDataLine line = (SourceDataLine)AudioSystem.getLine(info); line.open(stream.getFormat()); line.start(); runSonic(stream, line, speed, pitch, rate, volume, emulateChordPitch, quality, sampleRate, numChannels); line.drain(); line.stop(); } }
3、变调
package com.netmarch; /* This file was written by Bill Cox in 2011, and is licensed under the Apache 2.0 license. */ import com.google.common.io.ByteArrayDataOutput; import com.google.common.io.ByteStreams; import com.netmarch.web.open.util.Sonic; import com.sun.media.sound.WaveFileWriter; import javax.sound.sampled.AudioFileFormat; import javax.sound.sampled.AudioFormat; import javax.sound.sampled.AudioInputStream; import javax.sound.sampled.AudioSystem; import javax.sound.sampled.DataLine; import javax.sound.sampled.LineUnavailableException; import javax.sound.sampled.SourceDataLine; import javax.sound.sampled.UnsupportedAudioFileException; import java.io.ByteArrayInputStream; import java.io.File; import java.io.FileOutputStream; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; public class TestSonic { static ByteArrayDataOutput out = ByteStreams.newDataOutput(); // Run sonic. private static void runSonic( AudioInputStream audioStream, SourceDataLine line, float speed, float pitch, float rate, float volume, boolean emulateChordPitch, int quality, int sampleRate, int numChannels) throws IOException { Sonic sonic = new Sonic(sampleRate, numChannels); int bufferSize = line.getBufferSize(); byte inBuffer[] = new byte[bufferSize]; byte outBuffer[] = new byte[bufferSize]; int numRead, numWritten; sonic.setSpeed(speed); sonic.setPitch(pitch); sonic.setRate(rate); sonic.setVolume(volume); sonic.setChordPitch(emulateChordPitch); sonic.setQuality(quality); do { numRead = audioStream.read(inBuffer, 0, bufferSize); if(numRead <= 0) { sonic.flushStream(); } else { sonic.writeBytesToStream(inBuffer, numRead); } do { numWritten = sonic.readBytesFromStream(outBuffer, bufferSize); if(numWritten > 0) { line.write(outBuffer, 0, numWritten); byte[] target = new byte[numWritten]; System.arraycopy(outBuffer, 0, target, 0, numWritten); out.write(target); } } while(numWritten > 0); } while(numRead > 0); //saveFile(out.toByteArray()); } public static void saveFile(byte[] bytes) { String fileName = "c:/"+ System.currentTimeMillis() + ".wav"; OutputStream outStream = null; try { outStream = new FileOutputStream(new File(fileName)); WaveFileWriter writer = new WaveFileWriter(); AudioFormat frmt = new AudioFormat(16000, 16, 1, true, false); ByteArrayInputStream bi = new ByteArrayInputStream(bytes); InputStream in = new AudioInputStream(bi, frmt,bytes.length); writer.write((AudioInputStream) in, AudioFileFormat.Type.WAVE, outStream); outStream.close(); } catch (Exception e) { e.printStackTrace(); } } public static void main( String[] argv) throws UnsupportedAudioFileException, IOException, LineUnavailableException { float speed = 1.0f; float pitch = 1.500f; //大于1音调变高，小于1音调变低 float rate = 1.0f; float volume = 0.5f; boolean emulateChordPitch = false; int quality = 0; AudioInputStream stream = AudioSystem.getAudioInputStream(new File("d:\generated.wav")); AudioFormat format = stream.getFormat(); int sampleRate = (int)format.getSampleRate(); int numChannels = format.getChannels(); SourceDataLine.Info info = new DataLine.Info(SourceDataLine.class, format, ((int)stream.getFrameLength()*format.getFrameSize())); SourceDataLine line = (SourceDataLine)AudioSystem.getLine(info); line.open(stream.getFormat()); line.start(); runSonic(stream, line, speed, pitch, rate, volume, emulateChordPitch, quality, sampleRate, numChannels); line.drain(); line.stop(); } }
当然你也可以同事设置 volume,speed,pitch
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